When the underlying voice media of a Contact Center is VoIP, an incoming customer call is received at a gateway. The incoming call is split into a session initiation protocol (SIP) portion and a real time protocol (RTP) portion, sent to a media server (VMS) and a workflow is started for that customer call. The SIP portion of the call contains control signaling to facilitate the routing of the call while the RTP portion of the call contains the actual voice data. In current VoIP networks, when the workflow determines that the customer call is to be on hold, the VMS provides an on-hold playback to the customer through the gateway until the workflow determines that the customer call is to be taken off hold. Because the on-hold playback requires bandwidth for each caller on hold for a continuous amount of time, this system can consume a large amount of bandwidth when many callers are on hold.
The problem with this type of system is apparent and magnified when the VoIP network is connected across large distances through the use of wide area networks (WANs). For example, when a VoIP network has a gateway in a first city, but the customer call received in that gateway is properly routed to a VMS in a second city through a WAN, the on-hold playback must be transmitted continuously through the WAN and the gateway to the caller. Because WAN bandwidth is expensive, and on-hold calls consume bandwidth while the on-hold playback is continuously playing, current systems are unnecessarily expensive and inefficient.